[Libav-user] g729 in rtp

2018-03-13 Thread lagavulin2016 

Hello. I'm trying to convert voip pcap to wav.voip pcap has a few bidirectional call and includes sip and rtp packets.(for example, https://wiki.wireshark.org/SampleCaptures#SIP_and_RTP)
So I created libavformat/voip_pcap.c which is similar to libavformat/rtsp.cwith pcap file parsing instead of networking.
It seems to work except g729 codec.
1. g729 codec is not recognized because in rtp_payload_types from libavformat/rtp.c   "G729" codec_id is AV_CODEC_ID_NONE. If I change to AV_CODEC_ID_G729, it is recognized well.   Is this intentionally none? or g729 in rtp is not supported?
2. g729 annex b has 2-bytes Silence Insertion Descriptor(SID) frame   but ffmpeg doesn't seem to support this.
Is there any plan to support g729(and annex b)? or am I doing something wrong?


___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user

Re: [Libav-user] g729 in rtp

2018-03-13 Thread Carl Eugen Hoyos 
2018-03-13 8:15 GMT+01:00, lagavulin2016 :
> Hello. I'm trying to convert voip pcap to wav.
> voip pcap has a few bidirectional call and includes sip and rtp packets.
> (for example, https://wiki.wireshark.org/SampleCaptures#SIP_and_RTP)
> So I created libavformat/voip_pcap.c which is similar to libavformat/rtsp.c
> with pcap file parsing instead of networking.
> It seems to work except g729 codec.
> 1. g729 codec is not recognized because in rtp_payload_types from
> libavformat/rtp.c
>"G729" codec_id is AV_CODEC_ID_NONE. If I change to AV_CODEC_ID_G729, it
> is recognized well.
>Is this intentionally none? or g729 in rtp is not supported?

I believe a patch to support G.729 over rtp would be very welcome.
Do you know how such a patch could be tested?

> 2. g729 annex b has 2-bytes Silence Insertion Descriptor(SID) frame
>but ffmpeg doesn't seem to support this.

Can you provide a real-life sample of G.729B?

Carl Eugen
___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user

Re: [Libav-user] h264 with SVC

2018-03-13 Thread Carl Eugen Hoyos 
2018-03-12 14:16 GMT+01:00, Vittalprasad :

> Does Latest FFMPEG supports h264 SVC extension?

Do you have a real-world sample?
Anybody actually using SVC?

Carl Eugen
___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user

[Libav-user] Encode PCM16 to AAC

2018-03-13 Thread Michael IV 
Hi. I have a PCM 16 bit (signed) stream,which I want to encode using AAC
codec.

Now I am following the setup of audio encoding example and where  I check
for
SAMPLE FMT support by the codec,the check fails:

   while (*p != AV_SAMPLE_FMT_NONE)
{
if (*p == c->sample_fmt)
{
break;
}
p++;
}

Because AAC codec which I opened contains only AV_SAMPLE_FMT_FLTP

Does it mean I can't use audio data as PCM16 to encode with AAC?

Thanks.
___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user

Re: [Libav-user] Encode PCM16 to AAC

2018-03-13 Thread Carl Eugen Hoyos 
2018-03-13 17:16 GMT+01:00, Michael IV :
> Hi. I have a PCM 16 bit (signed) stream,which I want to encode using AAC
> codec.
>
> Now I am following the setup of audio encoding example and where  I check
> for
> SAMPLE FMT support by the codec,the check fails:
>
>while (*p != AV_SAMPLE_FMT_NONE)
> {
> if (*p == c->sample_fmt)
> {
> break;
> }
> p++;
> }
>
> Because AAC codec which I opened contains only AV_SAMPLE_FMT_FLTP
>
> Does it mean I can't use audio data as PCM16 to encode with AAC?

Yes, either use the aresample filter to convert from pcm_s16le to pcm_flt
or use libswresample directly.

Carl Eugen
___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user

Re: [Libav-user] g729 in rtp

2018-03-13 Thread lagavulin2016 
Previous post was somewhat html-formatted. I re-posted.
> 2018-03-13 8:15 GMT+01:00, lagavulin2016 :
> > Hello. I'm trying to convert voip pcap to wav.
> > voip pcap has a few bidirectional call and includes sip and rtp packets.
> > (for example, https://wiki.wireshark.org/SampleCaptures#SIP_and_RTP)
> > So I created libavformat/voip_pcap.c which is similar to libavformat/rtsp.c
> > with pcap file parsing instead of networking.
> > It seems to work except g729 codec.
> > 1. g729 codec is not recognized because in rtp_payload_types from
> > libavformat/rtp.c
> >"G729" codec_id is AV_CODEC_ID_NONE. If I change to AV_CODEC_ID_G729, it
> > is recognized well.
> >Is this intentionally none? or g729 in rtp is not supported?
> 
> I believe a patch to support G.729 over rtp would be very welcome.
> Do you know how such a patch could be tested?
  Well..my demuxer is not ready to commit yet... but simple question is..
  AV_CODEC_ID_NONE means g729 over rtp is not supported? or codec is recognized elsewhere?
  Because g729 is very common in voip, so I couldn't believe ffmpeg doesn't support g729 over rtp.
  If it is really not supported, I would start thinking contribute.
> 
> > 2. g729 annex b has 2-bytes Silence Insertion Descriptor(SID) frame
> >but ffmpeg doesn't seem to support this.
> 
> Can you provide a real-life sample of G.729B?
> 
> Carl Eugen
  g729a pcap sample is in the wireshark sample page I posted, but g729b is hard to find on the internet :(
  I'll search more.


___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user

Re: [Libav-user] g729 in rtp

2018-03-13 Thread lagavulin2016 

> 2018-03-13 8:15 GMT+01:00, lagavulin2016 :> > Hello. I'm trying to convert voip pcap to wav.> > voip pcap has a few bidirectional call and includes sip and rtp packets.> > (for example, https://wiki.wireshark.org/SampleCaptures#SIP_and_RTP)> > So I created libavformat/voip_pcap.c which is similar to libavformat/rtsp.c> > with pcap file parsing instead of networking.> > It seems to work except g729 codec.> > 1. g729 codec is not recognized because in rtp_payload_types from> > libavformat/rtp.c> >    "G729" codec_id is AV_CODEC_ID_NONE. If I change to AV_CODEC_ID_G729, it> > is recognized well.> >    Is this intentionally none? or g729 in rtp is not supported?> > I believe a patch to support G.729 over rtp would be very welcome.> Do you know how such a patch could be tested?
  Well..my demuxer is not ready to commit yet... but simple question is..
  AV_CODEC_ID_NONE means g729 over rtp is not supported? or codec is recognized elsewhere?
  Because g729 is very common in voip, so I couldn't believe ffmpeg doesn't support g729 over rtp.
  If it is really not supported, I would start thinking contribute.
> > > 2. g729 annex b has 2-bytes Silence Insertion Descriptor(SID) frame> >    but ffmpeg doesn't seem to support this.> > Can you provide a real-life sample of G.729B?> > Carl Eugen
  g729a pcap sample is in the wireshark sample page I posted, but g729b is hard to find on the internet :(
  I'll search more.


___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user